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Old 20th July 2011, 20:02   #11141  |  Link
tebasuna51
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Nope (use the search, this question was answered already in this thread).
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Old 21st July 2011, 04:51   #11142  |  Link
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I put in a search of "6.1" and it told there are no results. Searching is fun with a crippled search function. It said the same thing when attempting to search on DTS-MA 6.1.

Last edited by lostclusters; 21st July 2011 at 04:55.
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Old 21st July 2011, 06:28   #11143  |  Link
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Use Google with search functions

Quote:
dts-ma 6.1 site:forum.doom9.org
Most search engines in forums don't allow you to search things less than 4 characters.
"quotes" are usually required.

Google has a better chance, since it's diverse.
dts-ma will show you dts-hd ma, dtsma, dts ma, etc with Google.
With the forum search, it looks for exactly what you type in.
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Old 22nd July 2011, 04:10   #11144  |  Link
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That was a very good tip, TYVM!
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Old 26th July 2011, 04:55   #11145  |  Link
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quick question:

If I transcode DTS-MA to FLAC, and later decide to transcode that FLAC back to DTS-MA with Surcode, will the end file be basically identical to the first, or will the -core or other information be lost forever?
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Old 26th July 2011, 21:43   #11146  |  Link
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Is it possible to disable that annoying "Pling" sound when done with a file?
And is it possible to skip the 1-pass wav file and encode directly to in example a m4a aac file or is that wav file needed?
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Old 26th July 2011, 21:55   #11147  |  Link
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Quote:
Originally Posted by mastrboy View Post
Is it possible to disable that annoying "Pling" sound when done with a file?
Rename or delete the file "success.wav" in your eac3to directory.

Quote:
Originally Posted by mastrboy View Post
And is it possible to skip the 1-pass wav file and encode directly to in example a m4a aac file or is that wav file needed?
I haven't tried, but adding "-no2ndpass" to your commandline might do it. The wav is written so that if a second pass is necessary, which isn't always known before the first pass is complete, you don't need to decode the audio again. You might have audiable problems if you use that option.
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Old 2nd August 2011, 12:03   #11148  |  Link
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I'm having trouble converting the following stream to LPCM:

DTS-HD Master Audio English 1064 kbps 1.0 / 48 kHz

The resultant LPCM file is playing way too fast. I take it because there's no official LPCM 1.0? In any case, how do I convert this to a normal, playable, LPCM track?

Thanks.
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Old 2nd August 2011, 23:13   #11149  |  Link
iSeries
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About the eac3to 6.1 DTS-HD downmixing 'bug', if I output to individual WAV files could someone let me know how to mix the back center channel with the surround left/right channels manually/correctly please?

Could I do it with eac3to, eg: eac3to rightsurround.wav+backcenter.wav output.wav

Last edited by iSeries; 2nd August 2011 at 23:22.
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Old 3rd August 2011, 10:56   #11150  |  Link
tebasuna51
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Nope, you can use an audio editor (like Audacity or others), a command line tool like sox, or Avisynth (with BeHappy, wavi, ...) to do a mix like:

BL = 0.5858 x SL + 0.4142 x BC
BR = 0.5858 x SR + 0.4142 x BC

The recommended .avs' can be something like:

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.4142)
Normalize()

To obtain BL channel, and for BR channel:

sr = WavSource("D:\...\SideRigth.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sr, bc, 0.5858, 0.4142)
Normalize()
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Old 3rd August 2011, 18:06   #11151  |  Link
iSeries
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tebasuna51, thanks! The resulting .wav files seem to be 32 bit float though (from BeHappy), what do I need to add to the scripts to get 24 bit?

Thanks again!

Last edited by iSeries; 3rd August 2011 at 18:28.
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Old 3rd August 2011, 20:09   #11152  |  Link
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add:

Code:
ConvertAudioTo24bit()
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Old 3rd August 2011, 21:03   #11153  |  Link
iSeries
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Thanks guys!

...is it ok to reduce the bit depth of the new .wav's from 32 bit float back to 24 bit with eac3to? or is it more recommended to do this with BeHappy?

Last edited by iSeries; 3rd August 2011 at 21:16.
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Old 3rd August 2011, 22:55   #11154  |  Link
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Also, do I need to normailize ALL the wav's that will make up the 5.1, or just the two created with BeHappy?
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Old 3rd August 2011, 23:54   #11155  |  Link
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Well, nothing is perfect. You can't obtain always the same global audio volume with 7 speakers than with 6.

If you want preserve the balance volume betwen front and back channels maybe you need make this mix:

BL = SL + 0.7071 x BC
BR = SR + 0.7071 x BC

preserve the 32 bit float and check if there are values greater than 0 dB, if yes you need use the normalize only over BL-BR and lose some back volume, if not you don't need the normalize and the balance volume is ok.
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Old 4th August 2011, 00:05   #11156  |  Link
iSeries
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Thanks, like this?

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.7071)
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Old 4th August 2011, 10:34   #11157  |  Link
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Quote:
Originally Posted by iSeries View Post
Thanks, like this?

sl = WavSource("D:\...\SideLeft.wav")
bc = WavSource("D:\...\BackCenter.wav")
MixAudio(sl, bc, 0.5858, 0.7071)
Almost correct but the SL / SR channels must be factored by 1, not 0.5858.
Replace your last line with this:
Code:
MixAudio(sl, bc, 1.0, 0.7071)
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Old 4th August 2011, 14:29   #11158  |  Link
iSeries
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Thanks! Sorry one last question lol, how can I check if clipping has occured? Am I right in thinking that if I open the wav in Audacity and nothing goes beyond 1.0 or -1.0 then there is no clipping?

Also, I don't really understand any of this so a brief explanation on the difference between

MixAudio(sl, bc, 0.5858, 0.4142)

and

MixAudio(sl, bc, 1.0, 0.7071)

...would be greatly appreciated

Last edited by iSeries; 4th August 2011 at 15:11.
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Old 4th August 2011, 15:58   #11159  |  Link
tebasuna51
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Quote:
Originally Posted by iSeries View Post
..., how can I check if clipping has occured? Am I right in thinking that if I open the wav in Audacity and nothing goes beyond 1.0 or -1.0 then there is no clipping?
You can use Audacity -> View -> Show clipping

MixAudio(sl, bc, 0.5858, 0.4142)
make:
BL = 0.5858 x SL + 0.4142 x BC
the audio never can be greater than 1.0 (0 dB), without clip

and

MixAudio(sl, bc, 1.0, 0.7071)
make:
BL = 1.0 x SL + 0.7071 x BC
the audio can be greater than 1.0, then you can have clipping.
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Last edited by tebasuna51; 4th August 2011 at 16:01.
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Old 4th August 2011, 17:26   #11160  |  Link
iSeries
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Many thanks. I used MixAudio(sl, bc, 1.0, 0.7071) for LOTR Fellowship 1/2, and this produced no clipping. The Two Towers 1/2 and Return of the King 1/2 did produce clipping using that so for these I re-did with MixAudio(sl, bc, 0.5858, 0.4142). I'm guessing thats all ok (Audacity shows no clipping for all wavs)
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